[SR-Users] sipp to kamailio doesn't match ACK [SOLVED]

[JR Richardson]

On Tue, Jun 15, 2010 at 3:44 PM, JR Richardson <jmr.richardson at gmail.com> wrote:
> Hi All,
>
> I have been in the lab testing various load balancing scenarios with
> kamailio 1.5 and 3.0 and distributing calls to asterisk servers.  I am
> running into an issue that I'm sure is related to sipp not kamailio or
> asterisk.  My testing is setup like this:
> sipp><kamailio stateful (dispatcher or carrierroute)><multiple asterisk servers
>
> My sipp scenario:
> sipp -sn uac 10.10.12.22 -i 10.10.14.97 -s 22265 -d 5000 -l 100 -r 1
> -trace_err -m 1 -nr
>
> What I'm seeing in my ngrep sip traces is the ACK coming back from
> sipp does not have the RR header or a proper ftag:
>
> Here is the Ok sent from kamailio 10.10.12.22 to sipp 10.10.14.97:
>
> U 10.10.12.22:5060 -> 10.10.14.97:5061
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-0.
> Record-Route: <sip:10.10.12.22;lr=on;ftag=10002SIPpTag001>.
> From: sipp <sip:sipp at 10.10.14.97:5061>;tag=10002SIPpTag001.
> To: sut <sip:22265 at 10.10.12.22:5060>;tag=as18ae7aa7.
> Call-ID: 1-10002 at 10.10.14.97.
> CSeq: 1 INVITE.
> User-Agent: Asterisk PBX.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> Supported: replaces.
> Contact: <sip:65 at 10.10.14.102>.
> Content-Type: application/sdp.
> Content-Length: 184.
> .
> v=0.
> o=root 23641 23641 IN IP4 10.10.14.102.
> s=session.
> c=IN IP4 10.10.14.102.
> t=0 0.
> m=audio 24270 RTP/AVP 0.
> a=rtpmap:0 PCMU/8000.
> a=silenceSupp:off - - - -.
> a=ptime:20.
> a=sendrecv.
>
>
> Here is the reply ACK sent from sipp 10.10.14.97 to kamailio 10.10.12.22:
>
> U 10.10.14.97:5061 -> 10.10.12.22:5060
> ACK sip:22265 at 10.10.12.22:5060 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.14.97:5061;branch=z9hG4bK-10002-1-5.
> From: sipp <sip:sipp at 10.10.14.97:5061>;tag=10002SIPpTag001.
> To: sut <sip:22265 at 10.10.12.22:5060>;tag=as18ae7aa7.
> Call-ID: 1-10002 at 10.10.14.97.
> CSeq: 1 ACK.
> Contact: sip:sipp at 10.10.14.97:5061.
> Max-Forwards: 70.
> Subject: Performance Test.
> Content-Length: 0.
>
> And the ACK goes into the to-tag portion of the kamailio route script
> but can not match the transaction so kamailio exits, the dialog drops
> out of memory and when sipp sends a BYE, kamailio replys with 404 not
> found.
>
> So if I'm diagnosing this correctly, sipp is not maintaining the
> Record-Route: header in the responses back to kamailio and without
> that info kamailio can not maintain transaction state and the call
> fails.  Is there any work around or possibly another SIP performance
> testing suite that will handle RR headers?
>
> Thanks.
>
> JR
> --
> JR Richardson
> Engineering for the Masses
>
After reassuring myself that my diagnosis was correct and shortly
before I went blind tracing sip messages, I dug into sipp and found
the ability to create a new scenario adding the record route header
and next_url parameter.  So now when I run the testing from sipp, the
dialogs are properly processed statefully through kamailio and all
calls complete as expected.

I though I new SIP pretty well, but it's the problems I run into that
convince me there is so much more to learn.

Thanks.

JR
-- 
JR Richardson
Engineering for the Masses